[..] |
ASTERISK-19000: Multiple ChanSpy users cannot hear each other |
ASTERISK-19002: Using realtime sippeers: no value given for outbound proxy on line 0 of sip.conf |
ASTERISK-19003: asterisk closes TLS connection after receiving ACK for 401 following INVITE sent by Snom m9 |
ASTERISK-19004: RTP timeout won't work with locally bridged SIP channels |
ASTERISK-19005: Not working parameter mailcmd in voicemail.conf |
ASTERISK-19009: Deadlock on sip_new and load_realtime_queue. |
ASTERISK-19011: crashing res_odbc because of use of obj->con while reconnecting |
ASTERISK-19012: CLONE - [patch] CCSS: Sending a NOTIFY without the Subscription-State header |
ASTERISK-19013: T.38 port negotiation problem |
ASTERISK-19029: amaflags is not copied to channel for outgoing sip call |
ASTERISK-19030: Invalid host= declaration causes crash |
ASTERISK-19031: Asterisk can seg fault on invalid tcptls_session reference |
ASTERISK-19034: CLONE -[patch] New manager option enabledevents |
ASTERISK-19039: Indirect Ring Group Routing Not Ringing All Phones in Ring Group |
ASTERISK-19040: Asterisk 1.8.9.0 Blockers |
ASTERISK-19041: Asterisk 10.1.0 Blockers |
ASTERISK-19042: When joining ConfBridge, channel mutex can be free'd more times then it is locked |
ASTERISK-19048: T.38 Fallback to G.711 fails upon 503 response |
ASTERISK-19049: CDR wasn't generated after doing redirect through AGI |
ASTERISK-19050: Wrong transport for outgoing INVITE |
ASTERISK-19053: Investigate why the gateway_mix2 test is failing. |
ASTERISK-19055: Memory leaks in app_followme find_realtime |
ASTERISK-19056: Incorrect description for MESSAGE_SEND_STATUS variable in main.message.c |
ASTERISK-19057: [patch] message-summary NOTIFY: Port in Message-Account added twice and mwi_from (sip.conf) has no effect |
ASTERISK-19058: Messagesend and SIPFROMUSER |
ASTERISK-19061: Wiki Documentation on Realtime Database Connector possible incorrect syntax for extconfig.conf file. |
ASTERISK-19062: app_stack: cannot access memory at address 0x0 |
ASTERISK-19063: The channel fall on pickup with option sendrpid=yes |
ASTERISK-19079: Asterisk will not build under Freebsd with GCC 4.6 installed |
ASTERISK-19081: Call files in /var/spool/asterisk/outgoing are sometime not read and processed by pbx_spool.c |
ASTERISK-19082: Forwarding voicemail generate error in multi-tenant configuration |
ASTERISK-19087: CLONE -core show channels randomly shows IP instead of IAX account |
ASTERISK-19088: CLONE -Implicit Assumption About Dynamic Features |
ASTERISK-19089: faxdetect=yes in sip.conf general overrides faxdetect=no in peer configuration |
ASTERISK-19091: ConfBridge cannot handle multiple menu DTMF selections in rapid succession |
ASTERISK-19092: cisco phone 79xx can't register |
ASTERISK-19093: sip reload not loading all users |
ASTERISK-19094: Incorrect -x command line parameter behavior |
ASTERISK-19095: REGRESSION after r336294: Music on hold when sip call is put on hold doesnt work anymore after 1.8.8.0-rc1 |
ASTERISK-19096: Allow specifying which MixMonitor to stop |
ASTERISK-19097: Click To Call Busy Destination Results In Hangup |
ASTERISK-19098: chan_mgcp being enabled without dependencies met |
ASTERISK-19099: ConfBridge, set marked ( to send video) from console. |
ASTERISK-19100: ConfBridge crashes on closing timer when destroying conference bridge |
ASTERISK-19103: When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used. |
ASTERISK-19104: transmit_refer doesn't send "Replaces" in tag "Refer-To:" |
ASTERISK-19105: [JABBER± Jitsi call cause Asterisk to SEGFAULT |
ASTERISK-19106: SIP registration fails after temporary dns failure |
ASTERISK-19107: #include configuration file if it exists (do not fail if it's missing) |
ASTERISK-19108: TESTTIME function replies errors |
ASTERISK-19109: [patch] "deaf" participant support in ConfBridge |
ASTERISK-19126: Cannot install asterisk on mac os lion |
ASTERISK-19127: Asterisk does not quit on SIGTERM |
ASTERISK-19128: Asterisk 1.8.10 Blockers |
ASTERISK-19129: Asterisk 10.2.0 Blockers |
ASTERISK-19133: Memory leak using asterisk T.38 to/from T.30 gateway |
ASTERISK-19135: Asterisk ended with exit status 134 |
ASTERISK-19136: crash in odbc when using app_voicemail |
ASTERISK-19137: patten matching wrong when issue "dialplan reload" |
ASTERISK-19138: CLI does not Honor key bindings or keybinding file ~/.editrc |
ASTERISK-19139: SIP regression in 1.8 branch |
ASTERISK-19140: Put DAHDISpan and DAHDIChannel on some AMI events |
ASTERISK-19141: pub_lua Extensions and Execution of CONNECTED_LINE_CALLER_SEND_MACRO |
ASTERISK-19142: manager parameter channelvars=CHANNEL(dahdi_span) causes segmentation fault on Hangup |
ASTERISK-19143: Core dump when adding dialplan extension |
ASTERISK-19153: [patch] - Sms sender is not parsed correctly in incoming sms |
ASTERISK-19154: huge number of sip OPTION on 'sip reload' |
ASTERISK-19155: Memory leak in app_voicemail.c when using IMAP |
ASTERISK-19156: 1.8 SVN: channel.c:1474 __ast_queue_frame: Exceptionally long voice queue length queuing to Local during paging |
ASTERISK-19157: Failed to authenticate on INVITE to '"Anonymous" |
ASTERISK-19159: Asterisk fails to start MOH when SDP specifies connection IP of 0.0.0.0 only |
ASTERISK-19161: [patch] Add function REGISTRANT() that retrieves the peer that auto-registered an extension |
ASTERISK-19163: Got SIP response 400 "Bad Request" after Hangup |
ASTERISK-19164: ForkCDR with 'e' option to set end time is overzealous |
ASTERISK-19165: Empty CDR userfield and wrong UniqueID values stored in Master.csv when using Originate from AMI |
ASTERISK-19166: Retransmitted REGISTER requests are rejected with 401 (stale=true). |
ASTERISK-19167: Fix skipped tests in Asterisk Test Suite |
ASTERISK-19169: [patch] CallerID send before ring problem detected in chain_dahdi.c |
ASTERISK-19170: realtime queues fail to load queue information when there arent valid queue_members in the queue_members table |
ASTERISK-19171: sip tcp fails with secret |
ASTERISK-19172: Inconstistency for realtime colmn lastms |
ASTERISK-19173: All blind transfers failing on 1.8.9.0-rc1 |
ASTERISK-19176: The 'w' modifier support for ISDN spans was lost when sig_pri.c was extracted from chan_dahdi.c. Dial(DAHDI/g0/1234w888) |
ASTERISK-19178: Asterisk 10 beta2 disconnects on reload |
ASTERISK-19179: RTP inactivity SIP / ooh323 wont work |
ASTERISK-19180: ast_cel_fabricate_channel_from_event causes AMI VarSet events to be sent for a temporary/dummy channel |
ASTERISK-19181: SIP-Provider without "SIP/2.0 180 Ringing" no Audio when connected to DAHDi |
ASTERISK-19182: Crash in ast_channel_get_full() with partial name |
ASTERISK-19183: (Sporadically) missing connectedline event to caller channel in directed pickup app |
ASTERISK-19184: Crash at attempt to attended transfer a call |
ASTERISK-19186: Func_CURL is missing from wiki.asterisk.org, 1.8 functions section |
ASTERISK-19188: asterisk crashes if there no confbridge-join file |
ASTERISK-19189: AEL Macro and AELSub functions do not pass EXTEN variable, breaking CDR destination field |
ASTERISK-19190: AJAM Digest missing session cookie |
ASTERISK-19191: In AMI Redirect action somtimes channels get hangup while redirecting channels to meet me room |
ASTERISK-19192: ERROR we couldn't allocate a port for RTP instance |
ASTERISK-19193: Asterisk ended with exit status 134 |
ASTERISK-19196: Queue and local channels - Agent hunting order incorrect |
ASTERISK-19197: Calls from VOIP to Dahdi requiring transcoding fail |
ASTERISK-19198: Parallel make jobs break build |
ASTERISK-19199: Neither MATH nor $[] expression have the ABS(X) (absolute value) |
ASTERISK-19200: Not work alwaysauthreject=yes |
ASTERISK-19201: TLS Manager Bind Port - random port - not configurable |
ASTERISK-19202: CSipSimple (trunk) crushes Asterisk 1.8.8.1 (openSuse) |
ASTERISK-19203: Resource leak in SIP/TCP |
ASTERISK-19204: Manager API opens on random port on reload, TLS address not loaded as set |
ASTERISK-19205: Most Unique pattern matching broken when trailing "-" is part of extension |
ASTERISK-19206: Segmentation fault: menuselect/nmenuselect menuselect.makeopts |
ASTERISK-19209: Attended transfer failes |
ASTERISK-19213: deadlock on bultin atxfer |
ASTERISK-19215: Segfault chan_sip originating call |
ASTERISK-19216: cdr_pgsql reload failure |
ASTERISK-19220: chan_sip deadlock |
ASTERISK-19221: asterisk process hangs |
ASTERISK-19222: dialplan add extension documentation issue |
ASTERISK-19223: Called party keeps ringing until calling party has send a cancel |
ASTERISK-19231: Abort signal 6 raises when using 'sip show peers' with realtime peers |
ASTERISK-19232: Notifycid sending -1 instead of 1 |
ASTERISK-19233: patch to fix inband DTMF in chan_ooh323 |
ASTERISK-19234: Asterisk changes "From" header to "asterisk" when CALLERID(num-pres)=prohib_passed_screen is set |
ASTERISK-19235: confbridge fails: chan_sip.c:6544 sip_write: Can't send 10 type frames with SIP write |
ASTERISK-19240: UnParkedCall event does not contain the related parking lot name |
ASTERISK-19241: Cannot compile with Embedded Modules |
ASTERISK-19242: AMI QUEUESTATUS not working correctly in 10.0.1 (CLI queue show not working correct as well) |
ASTERISK-19243: DEVICE_STATE not correct when in h extension |
ASTERISK-19244: g729 not offered in SIP INVITE |
ASTERISK-19245: Fresh install of 1.8.9.0-rc3 hangs during module load and pegs one processor at 100% |
ASTERISK-19246: possible bug: Audiohook flag values overlap |
ASTERISK-19247: Delaying destroy of SIP INVITE dialog fail while call is allready Bridged. |
ASTERISK-19249: AMI PauseMonitor or UnpauseMonitor With Missing or Unknown Channel Forcibly Disconnects AMI Session. |
ASTERISK-19250: --enable-dev-mode should also apply to editline |
ASTERISK-19251: Manager eventq fills up with events with Usecount neq 0 |
ASTERISK-19252: qualify for h323 |
ASTERISK-19254: When working in real time with ARA and MySQL the backslashes not works properly |
ASTERISK-19264: ASTERISK-19202 creates: trap invalid opcode ip:516aa2 sp:7fff224a4640 error:0 in asterisk[400000+196000] on x86_64 builds |
ASTERISK-19265: ESwitch not Converting Variables |
ASTERISK-19266: flood of SQL warnings on 1.8 - 10.1 upgrade |
ASTERISK-19267: RSA key for TLS should not be stored in same file as cert |
ASTERISK-19268: Need to specify TLS peer verification policy per-peer |
ASTERISK-19270: CallerID missing on local channel |
ASTERISK-19271: Asterisk 1.8.11.0 Blockers |
ASTERISK-19272: Asterisk 10.3.0 Blockers |
ASTERISK-19273: Store Asterisk Manager HTTP Sessions in persistent storage. |
ASTERISK-19276: Google Calendar periodic event miss updating |
ASTERISK-19277: [patch]endlessly repeating error: "poll failed: Bad file descriptor" |
ASTERISK-19279: Asterisk stops processing Local channels - CLI is full of messages "Exceptionally long voice queue length queuing to Local/XXX" |
ASTERISK-19281: "sip show peers" show incorrect columns |
ASTERISK-19282: Add F option to Bridge command |
ASTERISK-19283: Add F option to Queue command (transfer on hangup) |
ASTERISK-19285: [regression] Deadlock in asterisk 1.8.9.0 (possible chan_agent and queues interaction) |
ASTERISK-19289: chan_iax2.so: undefined symbol: ast_aes_set_encrypt_key |
ASTERISK-19290: Voicemailmain password not recognized from Aastra 480i phone in versions past 10.0.1 |
ASTERISK-19291: Background in realtime |
ASTERISK-19292: New "dialplan remove context" and modification of "dialplan add include" |
ASTERISK-19293: Got SIP response 400 "Missing Subscription-State header" |
ASTERISK-19294: Asterisk 1.8.6.0 failed to switch RTP destination when receiving a SIP reinvite |
ASTERISK-19295: Segfault on "sip show peers" on Solaris |
ASTERISK-19296: Attended transfer and hangup events |
ASTERISK-19297: Call from 'SIPX' to extension 'xxxxxxxxx' rejected because extension not found in context 'from-SIPX' |
ASTERISK-19298: segmentation fault in chan_ooh323 |
ASTERISK-19299: [patch] AgentLogin Option To Skip Password Prompt |
ASTERISK-19300: chan_skype can not load under 1.8.9.0 |
ASTERISK-19301: ooh323 trunk to AVAYA |
ASTERISK-19302: Messagesend |
ASTERISK-19303: Asterisk does not acknowledge the ACK received to terminate the dialog. |
ASTERISK-19304: New feature to send udptl packets directly between both call legs |
ASTERISK-19305: After reciving INVITE with FROM user without phone number asterisk crashes with segfault |
ASTERISK-19306: Invalid parameters in rt_handle_member (app_queue.c: create_queue_member: No location at interface '') |
ASTERISK-19307: When a jabber server is configured as type=component, asterisk crashes with segmentation fault. |
ASTERISK-19308: problem with transit calls ooh323-dahdi(pri)-panasonic 500 |
ASTERISK-19309: [patch] DUNDi message routing bug |
ASTERISK-19310: 'i' option is defined twice at AST_APP_OPTIONS macro in app_page.c |
ASTERISK-19311: ParkAndAnnounce crash asterisk |
ASTERISK-19312: No DTMF decoding on outbound call via SS7 E1 channel |
ASTERISK-19313: [patch] incorrect handling of UPDATE response with canreinvite=update |
ASTERISK-19315: Impossibly High Lagged Value (Asterisk 1.8.8.1) |
ASTERISK-19316: Asterisk cannot detect canceled calls on analog lines |
ASTERISK-19317: QueueLog does not log ringnoanswer if the caller abandons while the agent's extension is ringing |
ASTERISK-19318: Asterisk locks up during Page cmd |
ASTERISK-19319: [patch] Triggers dialplan actions when specific CONTROL_FRAMES are detected on a channel. |
ASTERISK-19320: SIGSEGV when starting within mgcp module |
ASTERISK-19321: Transfer application ignores port information |
ASTERISK-19322: Polycom blind SIP transfer to park extension plays parking orbit number prompt to caller extension after transfer. |
ASTERISK-19332: chan_usbradio fails to compile under Ubuntu natty (--enable-dev-mode) |
ASTERISK-19334: Adaptive CDR via ODBC driver can't handle UTF8-type fields in database |
ASTERISK-19335: MeetMeAdmin(confno,N) mutes admins |
ASTERISK-19336: h exten is not run in the context that calls a AEL macro |
ASTERISK-19337: app_voicemail fails to compile with imap storage |
ASTERISK-19340: [patch] CALLERID(subaddr) only allows ASCII |
ASTERISK-19341: Missing initialization on bind_addr |
ASTERISK-19342: Initialize parking hints with NOT_INUSE state |
ASTERISK-19346: Registering multiple Google calendars via caldav crashes Asterisk |
ASTERISK-19347: File descriptor errors from res_timing_timerfd |
ASTERISK-19348: With alwaysauthreject=yes AND allowguest=no Asterisk fails to report a SIP Security Event |
ASTERISK-19350: [Regression] Asterisk realtime via ODBC - null is inserted in brackets '()' |
ASTERISK-19351: MeetMe does not record, WARNING: file.c:1168 ast_writefile: No such format '' |
ASTERISK-19352: SIP warning message if only UDP is eanbled |
ASTERISK-19353: musiconhold of remote party is replaced by local moh |
ASTERISK-19354: ConfBridge does not close channel when using local channels |
ASTERISK-19355: Call transfer with consultation frequently fails in cross-linked asterisk scenario (directmedia & sendrpid active) |
ASTERISK-19356: Deadlock in cel_sqlite3_custom module reload |
ASTERISK-19358: Failed to CANCEL a call in ringing state (SIP) in 1.8.9.2 |
ASTERISK-19360: When using DialPlan in RealTime, TIMEOUT(response) does not work |
ASTERISK-19361: Asterisk exited on signal 6: Related to sip show peers? |
ASTERISK-19362: Loud squelch after conf-hasjoin: "is now in the conference" announcement |
ASTERISK-19363: DAHDI PRI channel becomes unusable after a while, and calls fail over it. |
ASTERISK-19364: add concise option to core show hints |
ASTERISK-19365: Remote SIP Call legs are frequently not released in a cross-linked Asterisk scenario (directmedia & sendrpid) |
ASTERISK-19366: Periodic RTCP receiver reports in cross-linked asterisk scenario although asterisk is no longer in the RTP path (directmedia) |
ASTERISK-19367: Update Debian Install Prerequisite install |
ASTERISK-19368: Queue penalty only work when QUEUE_MIN_PENALTY == QUEUE_MAX_PENALTY |
ASTERISK-19369: SIP timeout to peer causes hangup, call does not continue to next priority. |
ASTERISK-19370: format_ogg_vorbis fail to compile |
ASTERISK-19371: Incorrect matching with new pattern match engine enabled |
ASTERISK-19372: BUSY/INCOMPLETE/CONGESTION indications not passed to SS7 channel |
ASTERISK-19373: Segmentation Fault in ast_udptl_write() due to bad memcpy() call |
ASTERISK-19374: No audio in Gtalk calls |
ASTERISK-19379: IAX channel chooses the wrong password for authentication |
ASTERISK-19380: Asterisk 10.2.0-rc2 MessageSend() application reply to sender issue |
ASTERISK-19381: Local channel don't inherited language |
ASTERISK-19382: Park() ignores 'r' option, plays default MOH instead. |
ASTERISK-19383: Asterisk 1.8.5.0 - atxfer authorization problem when a call returns for reject or no answer |
ASTERISK-19384: REGRESSION - CLONE - CDR(accountcode) not accessable to 'Local' channels |
ASTERISK-19385: "Callerid:" in call-files and Asterisk Manager doesn't work |
ASTERISK-19386: Channel group not released after a channel has been Chanspy'd upon |
ASTERISK-19387: Seg Fault upon Asterisk Startup |
ASTERISK-19388: Make it possible to put any connected call on hold, not just bridged ones |
ASTERISK-19389: Sending ACK in CANCLE dialog to the wrong destination |
ASTERISK-19391: Unable to edit From Caller Name with RPID |
ASTERISK-19397: Fix cause code for no channel available |
ASTERISK-19407: Set CDR variable ignored on record created after after ForkCDR |
ASTERISK-19409: TestSuite: twisted reactor incompatible with python subprocess module |
ASTERISK-19411: Conference Participants are placed back on hold when marked user quits |
ASTERISK-19416: H323 trunking failure. |
ASTERISK-19417: CLONE - Unable to edit From Caller Name with RPID |
ASTERISK-19418: Silence Suppression with TDM cards |
ASTERISK-19419: Asterisk does not compile under dev mode with gcc 4.6.3 |
ASTERISK-19420: Segfault in chan_mgcp |
ASTERISK-19421: app_rpt cannot be compiled with --enable-dev-mode (ubuntu 11.10+) |
ASTERISK-19422: CCSS does not function if "sip" is used instead of "SIP" when dialing |
ASTERISK-19423: Issue regarding CDR_ADAPTIVE_ODBC.c versus CDR_ODBC.c |
ASTERISK-19424: Spurious hangups during ringing on analog DAHDI channels |
ASTERISK-19425: Calls not released after BYE |
ASTERISK-19426: Mixmonitor does not create file and record anything |
ASTERISK-19428: Confbridge allows more max_members than set |
ASTERISK-19429: ERROR[23303]: astobj2.c:110 INTERNAL_OBJ: user_data is NULL when login to AMI |
ASTERISK-19430: 1.8.9.1 SIP NOTIFY crashes 2wire (U-Verse) routers |
ASTERISK-19431: Asterisk Russian language support missing voicemail prompts |
ASTERISK-19432: Seg Fault in libresample upon Asterisk Startup |
ASTERISK-19433: CLONE - Called party keeps ringing until calling party has send a cancel |
ASTERISK-19434: Segmentation fault on starting ConfBridge |
ASTERISK-19435: Asterisk segfaults in app_alarmreceiver |
ASTERISK-19436: outbound fax over t38 gateway can't pass |
ASTERISK-19440: ConfBridge does not turns off MOH when participant kicked out |
ASTERISK-19441: menuselect makes <use> tags shown and executes as "Depends on:" |
ASTERISK-19442: unaccepted attend transfer hangup caller |
ASTERISK-19443: realtime peers are not loaded during start |
ASTERISK-19444: Usage for CLI command 'devstate change' is truncated by an unnecessary comma |
ASTERISK-19445: Incorrect values are specified as length in memcpy and memset |
ASTERISK-19446: Improvement to MWI (with Teksavvy Tektalk service (Metaswitch Networks equipment) |
ASTERISK-19447: [patch] Add IPv6 Address Support To Security Events Framework |
ASTERISK-19448: ConfBridge crashes Asterisk when no timing module loaded. |
ASTERISK-19449: Can't connect to asterisk's console using command asterist -r |
ASTERISK-19450: there always an extra byte added to contact header while send "ACK" request. |
ASTERISK-19451: va_start/va_copy and va_end do not always match up |
ASTERISK-19452: ChanSpy with MixMonitor test sporadically fails |
ASTERISK-19454: outbound proxy not being cleared which sip reload performed |
ASTERISK-19455: SIP channels permanently stuck in system after BYE message received |
ASTERISK-19456: Turn Off Warning Message When Bind Address Is Set To ANY |
ASTERISK-19457: Re-add macro option for stdexten to support legacy dialplans |
ASTERISK-19459: Asterisk sending BYE on wrong NIC |
ASTERISK-19460: [patch] Function TXTCIDNAME never actually makes DNS calls and always returns an empty string |
ASTERISK-19461: ChanSpy - Improper refcounts avoid channel release |
ASTERISK-19462: asterisk Illegal Instruction (core dumped) |
ASTERISK-19463: Asterisk deadlocks during startup with mutex errors |
ASTERISK-19465: P-Asserted-Identity Privacy |
ASTERISK-19466: NSGate 39xx series gateway will not register with pedantic=yes |
ASTERISK-19467: Error should always be logged if SIP message fails compliance check |
ASTERISK-19468: Crash seems to be related to sip fax calls |
ASTERISK-19469: Execution of audio playbacks from call time limits (Dial application 'L' option) do not reflect configured options |
ASTERISK-19470: Documentation on app_amd is incorrect |
ASTERISK-19471: ConfBridge does not record anything |
ASTERISK-19483: Hangup during Read() or Record() causes hangup extension failure |
ASTERISK-19487: AMI module reload causes deadlock |
ASTERISK-19488: Rejected supervised transfer hangs up on calling party |
ASTERISK-19491: MeetMe fails to remove participant from conference when user leaves during playback of 'conf-onlyperson.ulaw' |
ASTERISK-19492: Group write permission removed from existing directory /etc/asterisk/. when updating |
ASTERISK-19493: ChanSpy onto a Local channel can leave a hung channel |
ASTERISK-19494: Blind Transfer to Park extension does not set timeout destination correctly on 2nd Park attempt. |
ASTERISK-19495: Create CEL tests for the Asterisk Test Suite |
ASTERISK-19497: ConfBridge recording does not work reliably |
ASTERISK-19498: All calls via IAX2 fail |
ASTERISK-19499: ConfBridge MOH is not working for transferee after attended transfer |
ASTERISK-19500: T.38 session fails with '488 Not acceptable here' if within 5 seconds on asterisk 1.8 |
ASTERISK-19501: Channel group on a Local channel not released after it has been Chanspy'd upon |
ASTERISK-19502: Wrong port specified on SIP INVITE response when using custom TCP port |
ASTERISK-19503: Aastra 480i loses mwi light after every reboot until reloading Asterisk. |
ASTERISK-19504: Queue Memeber Penalty not considered if queue reloaded after penalty modifieing |
ASTERISK-19505: Crash during high usage when Dial Time out set to 280 |
ASTERISK-19506: LIMIT_WARNING_FILE plays warning to both participants BUT one after another, not at the same time |
ASTERISK-19508: res_srtp.so crash with snom phone 370 on srtp_unprotect_rtcp |
ASTERISK-19510: RTP stream/Directmedia/Musiconhold/SIP not working |
ASTERISK-19511: Dial I option ignored if dial forked and one fork redirects |
ASTERISK-19512: force rport inconsistent between sip show peer and peers |
ASTERISK-19513: app_voicemail fails to compile with IMAP storage |
ASTERISK-19514: Queue realtime does not remove members after removed from database: Queue fraud has no realtime members defined. No need for update |
ASTERISK-19515: Need hooks for resource to leverage for NAT hole poking for media streams |
ASTERISK-19516: [patch] Enable RFC 4662/Broadsoft Resource list subscriptions in Asterisk 1.8.9.0 |
ASTERISK-19518: Voice Mail with IMAP storage reports erroneous error message in FreePBX configurations |
ASTERISK-19519: [patch] valid IP address in RTP offer when Asterisk is attached to several networks |
ASTERISK-19520: Call IAX2 - PRI strong distortion when using 16kHz speex |
ASTERISK-19521: chan_iax2 does not honor trunkfreq config option |
ASTERISK-19522: realtime peers are not loaded during start |
ASTERISK-19531: Realtime SIP peers that explicitly unregister have incorrect device state. |
ASTERISK-19532: Asterisk crashed after connecting with jabber server in component mode |
ASTERISK-19533: Script run from #exec can't connect to the manager on initial start or restart of Asterisk |
ASTERISK-19535: Dial/queue should handle HOLD/UNHOLD control frame similar to connected line updates. |
ASTERISK-19536: Queue option ringinuse is ignored |
ASTERISK-19537: Deadlock potential in ast_do_masquerade() because it calls ast_indicate with the channel lock held. |
ASTERISK-19538: Asterisk segfaults on sippeers realtime redundancy |
ASTERISK-19539: jabber outgoing messages become incomings |
ASTERISK-19540: Use of GNU old-style field designator extension |
ASTERISK-19541: Security Vulnerability: remotely exploitable stack overrun in Milliwatt |
ASTERISK-19542: Security Vulnerability: remotely exploitable stack overflow in main/utils ast_parse_digest |
ASTERISK-19547: utils.c:1236 ast_careful_fwrite: fflush() returned error: Bad file descriptor |
ASTERISK-19548: Ability to run dialplan on callee channel before making call upon Dial() |
ASTERISK-19549: Channel Hangup Handlers |
ASTERISK-19550: Segfault in ast_readaudio_callback |
ASTERISK-19551: Dial with Gosub autoservice error message is misleading |
ASTERISK-19552: CDR logs on call transfers prints only last leg |
ASTERISK-19553: Cannot perform feature attended transfer after pickup when using PickupChan |
ASTERISK-19554: chan_unistim notes warnings about retransmissions of ACK and multiple ACKs received |
ASTERISK-19555: SIP name in RDNIS, not CallerID number |
ASTERISK-19556: Asteriskt thread use 99% cpu |
ASTERISK-19557: [Regression] Segfault in res_jabber.c |
ASTERISK-19558: SIP INVITE header is broken |
ASTERISK-19559: No sound in calls after 1-2 seconds (SIP to IAX2) |
ASTERISK-19560: broken CANCEL is record routes are added to ringing |
ASTERISK-19561: libwat support (Wireless AT Library) |
ASTERISK-19562: [patch] ConfBridge - Inconsistent hold-music behaviour |
ASTERISK-19565: Investigate failures of the nat_supertest in the Asterisk Test Suite |
ASTERISK-19567: Investigate module load / unload failures in dynamic_modules test in Asterisk Test Suite |
ASTERISK-19571: [patch][feature] ConfBridge - Support for playing back arbitrary messages to individuals or the whole bridge |
ASTERISK-19572: Proposed change to CDR MySQL table structure |
ASTERISK-19573: hangup is not detected during call to func_curl within pbx_lua when a SIP caller hangs up |
ASTERISK-19574: Directory application should set variable upon successful search. |
ASTERISK-19575: AMI Hangup channels by regex |
ASTERISK-19576: DTMF Passed Unreliably from DAHDI Analog to GTalk |
ASTERISK-19577: Overcoming 64 callgroup / pickupgroup limit by creating "group contexts" |
ASTERISK-19578: ERROR we couldn't allocate a port for RTP instance while DAHDI bridgeing |
ASTERISK-19579: ERROR we couldn't allocate a port for RTP instance while DAHDI bridgeing |
ASTERISK-19580: chan_gtalk crash Asterisk on outgoing calls |
ASTERISK-19590: call completed the agent fixes a completecaller |
ASTERISK-19591: CallCompletion atumatic cancel on call return |
ASTERISK-19592: Security Vulnerability: heap overflow exists in chan_skinny's handling of KEYPAD_BUTTON_MESSAGE |
ASTERISK-19594: app_meetme unable to write frame (stuck channel) |
ASTERISK-19595: Inefficient wav49 disk writes |
ASTERISK-19596: Memory Leak of 8gigs RAM. |
ASTERISK-19597: Failure to pass NULL data pointer with AST_CONTROL_HOLD frame causes crash when MOH is started |
ASTERISK-19598: Garbled audio using Page app and MulticastRTP channel |
ASTERISK-19599: Announce parking slot number to callee |
ASTERISK-19601: Failure of domain matching on authentication of INVITE request produces misleading NOTICE message; bypasses alwaysauthreject logic |
ASTERISK-19603: Asterisk AMI truncates long responses over medium latency connections |
ASTERISK-19604: pin parameter of meetme ignored for dynamically added conferences |
ASTERISK-19606: CLONE - Directory application should set variable upon successful search. |
ASTERISK-19608: Asterisk-1.8.x starts rejecting calls with cause code 44 after some time. |
ASTERISK-19609: SRTP to RTP bridging with two crypto lines in SDP does not work |
ASTERISK-19610: dsp.c can no longer detect a quick DTMF sequence |
ASTERISK-19611: SIP stack stops working (deadlock?) if a call to a snom phone is redirected by "302 Moved temporarily" to chan_local |
ASTERISK-19612: Asterisk crash with compiler flag LOW_MEMORY |
ASTERISK-19613: Multibytes characters in files are not handled properly (signed char compared to int will get incorrect result if this byte is one of multibyte character) |
ASTERISK-19614: ILBC 20ms force |
ASTERISK-19618: Asterisk 1.8.12.0 Blockers |
ASTERISK-19619: Asterisk 10.4.0 Blockers |
ASTERISK-19620: directrtpsetup is not working anymore as expected/as in earlier Asterisk versions |
ASTERISK-19628: Crash during blind transfer with chan_unistim |
ASTERISK-19632: Trouble with fax gateway |
ASTERISK-19633: Having any h extension in peer's context breaks unaccepted attended feature transfers |
ASTERISK-19634: Sending DTMF tones using the AMI through the agent proxy channel doesnt work as expected |
ASTERISK-19635: Hangup is always recorded in queue_log as COMPLETECALLER when 'h' extension is present |
ASTERISK-19636: Asterisk crashes during attended transfer due to bad data pointer passed in HOLD frame from chan_iax2 |
ASTERISK-19637: Userfield variable cannot be correctly set in h extension |
ASTERISK-19638: Problem: Asterisk wont start |
ASTERISK-19639: [patch] - Deadlock in queue with attended transfer |
ASTERISK-19640: aastra-xml UDP problem! |
ASTERISK-19641: ConfBridge app plays conf-placeintoconf message to bridge, and not to joining channel |
ASTERISK-19642: SIP channels permanently stuck in system after BYE message received |
ASTERISK-19643: codec_dahdi: Block on frameout if the hardware has enough samples to complete a frame. |
ASTERISK-19644: Loss of audio while running ChanSpy() on a specific prefix; channel gets stuck permanently. |
ASTERISK-19645: Asterisk doesn't respect the video codec order |
ASTERISK-19646: Fix typo \n in chan_sip SDP negotiation warning message |
ASTERISK-19647: talktime 0 sec after transfer queueA->QueueB |
ASTERISK-19651: Coverity Report: Fix issues for error type SIZEOF_MISMATCH |
ASTERISK-19655: Coverity Report: Fix issues for error type NEGATIVE_RETURNS |
ASTERISK-19662: Coverity Report: Fix issues for error type MISSING_BREAK |
ASTERISK-19665: Coverity Report: Fix issues for error type RESOURCE_LEAK |
ASTERISK-19668: Coverity Report: Fix issues for error type OVERRUN_STATIC |
ASTERISK-19671: Coverity Report: Fix issues for error type REVERSE_NEGATIVE |
ASTERISK-19675: MWI stops working after lifting handset or using phone |
ASTERISK-19676: Fax Sessions are not always reported by 'fax show sessions' |
ASTERISK-19677: SIP dial string //IPorHost does not work like expected |
ASTERISK-19678: Manager disconnects on return of large dataset with action_command |
ASTERISK-19680: Monitor application docs are missing/incorrect |
ASTERISK-19682: Parsing of XML document tag <variable> malforms wiki documentation |
ASTERISK-19684: [T.38 gateway] [irroot t38gateway-1.8 branch] I can receive faxes with t38 gateway, but send fails |
ASTERISK-19708: Call Deflection with DAHDISendCallreroutingFacility on EuroISDN not working |
ASTERISK-19709: fix so connectab takes the ser_dialog_custom table into account |
ASTERISK-19711: Crash emanating from add_exten_to_pattern_tree() |
ASTERISK-19712: Retrieve of fields from Calendar EWS |
ASTERISK-19713: Asterisk segfaults on invalid datastore in channel destructor |
ASTERISK-19714: No hungup after BYE. |
ASTERISK-19716: Don't validate Contact URI hostpart when nat=yes |
ASTERISK-19717: Attended transfer hangup |
ASTERISK-19718: ast_app_inboxcount2() calls ast_inboxcount2_func without checking if it's assigned (instead checks ast_inboxcount_func) |
ASTERISK-19719: Asterisk doesn't add Contact field in 200 OK when ACK for 401 response is out of order |
ASTERISK-19720: Using SIP realtime with caching incoming calls are routed to dialplan context specified in sip.conf general context and not the context specified in SIP peer context until peer is refreshed from database |
ASTERISK-19721: Asterisk core sounds, italian version |
ASTERISK-19722: Manager disconnects under high originate load volume |
ASTERISK-19723: Blind parking does not work anymore |
ASTERISK-19724: [patch][feature] ConfBridge - Support for enabling MoH playback in conferences |
ASTERISK-19725: [patch][feature] ConfBridge - Support for getting the name-recording file created when participants announce themselves |
ASTERISK-19726: [patch][bug] ConfBridge - Users listening to MoH, and who should be muted, are often unmuted and recorded |
ASTERISK-19727: MixMonitor does not work on local channels |
ASTERISK-19730: Stack overflow in chan_sip when destroying mwipvt |
ASTERISK-19734: Having any h extension in peer's context breaks unaccepted attended feature transfers |
ASTERISK-19738: Calendar EWS does not attempt to extract the Body element in a CalendarItem and populate the description event field |
ASTERISK-19748: Add LinkedID to AMI Events |
ASTERISK-19749: Calendar EWS can't force event trigger to execute dialplan context without adding channel to calendar.conf |
ASTERISK-19750: display_send=name_initial doesn't work |
ASTERISK-19751: asterisk fails to autoconfigure - cannot find sqlite3 |
ASTERISK-19752: Add a channel variable/configuration option for defining pre-recorded announcement when "announce_join_leave=yes" |
ASTERISK-19753: App Macro argument processing does not honor escaped, quoted, or nested parentheses commas. |
ASTERISK-19754: Deadlock in chan_sip / pthread_timing |
ASTERISK-19755: __ao2_ref() validates user_data twice |
ASTERISK-19756: Use of asprintf with MALLOC_DEBUG could corrupt memory or crash. |
ASTERISK-19758: main/asterisk.c rawmemchr() undefined on OpenBSD |
ASTERISK-19759: Missing Payload Type From Events API |
ASTERISK-19760: Update Security Events Unit Tests |
ASTERISK-19761: mp3_read crash |
ASTERISK-19762: Segfault in ast_frdup when invalid data length specified in duplicated frame |
ASTERISK-19763: Confbridge with SIP to Local channel results in hung video; Exceptionally long queue length queuing to Bridge |
ASTERISK-19764: Infinite loop with autoservice when looking for nonexistant extension label. |
ASTERISK-19765: CLONE - ACK is ignored upon call-pickup with sendrpid=yes |
ASTERISK-19766: Linkedid field is too short |
ASTERISK-19767: dead lock in res_config_pgsl while dialing SIP extension |
ASTERISK-19768: ConfBridge - sound_muted/sound_unmuted aren't playing |
ASTERISK-19769: main/asterisk.c rawmemchr() undefined on *BSD () |
ASTERISK-19770: Security Vulnerability: Segmentation fault when receiving an out-of-dialogue SIP UPDATE including a rpid info |
ASTERISK-19771: User is unable to customize sound_leader_has_left |
ASTERISK-19772: [branch] Making it possible to set minimum DTMF duration without patching channel.c |
ASTERISK-19773: Asterisk crash on issuing Asterisk-CLI 'reload' command multiple times on cli_aliases |
ASTERISK-19775: Backgrace generation in Asterisk causes a seg fault |
ASTERISK-19778: dialplan function MANAGER_ONLINE(username) |
ASTERISK-19779: Asterisk segfaults when handling sip_security_event and attempting to load realtime peer with no realtime backend |
ASTERISK-19780: Asterisk segfaults on invalid datastore in channel destructor |
ASTERISK-19793: Only last realtime member of a queue is not actually removed from queue when removed from database |
ASTERISK-19794: IAX channel won't hangup after musconhold stopped |
ASTERISK-19795: Pinequeue: Play queue prompts in the background |
ASTERISK-19796: race between ast_readaudio_callback and ast_closestream |
ASTERISK-19797: GET_HEADER(Proxy-Authorization) get nothing |
ASTERISK-19799: Apparent deadlock between ODBC Queue log and ODBC CDR |
ASTERISK-19801: Deadlock with masquerade and chan_iax |
ASTERISK-19802: Fails to set WRITE mode correctly, skips first file to be played when format is ULAW. |
ASTERISK-19803: asterisk cli no tab completion for removing queue member with MEMBER_NAME variable |
ASTERISK-19805: Race condition exists between hanging up Conference Bridge Channel and servicing the channel |
ASTERISK-19807: Create masquerade "super-test" for Asterisk Test Suite |
ASTERISK-19810: Asterisk 10: AGI 'record file' broken in SVN 364536 |
ASTERISK-19815: Crash in core show locks when BETTER_BACKTRACES is enabled |
ASTERISK-19817: Call recording stops when call is transferred |
ASTERISK-19818: Rework Asterisk Test Suite version parsing |
ASTERISK-19820: wrapuptime is intermittently disregarded for queue calls |
ASTERISK-19821: DTMF conversion SIP INFO to RFC2833 changes duration |
ASTERISK-19822: EWS Calendar Integration Problem |
ASTERISK-19825: Bridge stuck for several minutes after hangup - possible hangup control frame skip |
ASTERISK-19827: Asterisk crash, whenever mwi => pass:user:authuser@host:port/mailbox is set in sip.conf |
ASTERISK-19828: MESSAGE_SEND_STATUS set to SUCCESS despite response of 400 Bad Request |
ASTERISK-19829: MessageSend disregards the port when specified in the from option of MessageSend(to[,from]) |
ASTERISK-19830: Asterisk receives an SDP offer with "recvonly" for video media but Asterisk responds with "sendrecv" |
ASTERISK-19834: Memory leak caused by thread created by bridge_channel_join being neither joined nor detached |
ASTERISK-19835: use webrtc iILBC code for codec_iLBC |
ASTERISK-19836: Since change 325816 (T.38-gateway code) ReceiveFAX via T.38 immediately fail. always |
ASTERISK-19837: Asterisk crashing regularly in 1.8.11.1 due to memory corruption |
ASTERISK-19838: From Header has capital A in userpart Anonymous if CALLERID(pres)=unavailable, RFC uses lower case anonymous |
ASTERISK-19839: Not all hints are displaying status correctly |
ASTERISK-19840: Disable global atxfernoanswertimeout |
ASTERISK-19842: POTS flashhook transfer causes deadlock |
ASTERISK-19844: Broadvoice Got SIP response 503 Service Unavailable |
ASTERISK-19845: Unable to register to sip through external pbx. Call ID Changed |
ASTERISK-19846: sip users/peers not matched on incoming invite when there are multiple A records in DNS |
ASTERISK-19847: Allow ConfBridge actions to be executed on channels in a conference via AMI |
ASTERISK-19848: Deadlock in Asterisk in ast_parse_device_state / Dial |
ASTERISK-19849: Issue also occurs when 3 people using IAX2 join the same meetme conference. Error messages recur until asterisk is restarted |
ASTERISK-19851: Asterisk Crashes in chan_sip when failing to create ast_str in init_resp |
ASTERISK-19852: Call pickup does not work with notifycid=yes |
ASTERISK-19853: CDR(custom_field) field set in a DYNAMIC_FEATURE is reverted to the previous value when the call is terminated. |
ASTERISK-19854: freeze channels showing in core show channels |
ASTERISK-19856: Transfer is being denied when global allowtransfer=no, ignoring peer setting |
ASTERISK-19857: Explore directmedia re-INVITE improvements between multiple Asterisk instances |
ASTERISK-19859: cid_tag is not set according to the sip configuration anymore if get_rpid() != 0 |
ASTERISK-19860: Wrong CDR duration values because of dependency to CDR write time especially in cdr batch mode |
ASTERISK-19861: External MWI subscriptions: Asterisk not responding to auth request |
ASTERISK-19862: app_queue: Update Data of Queues (use queues as outbound calls container) |
ASTERISK-19863: CallCounter not utilized by app_queue - members in busy state |
ASTERISK-19864: Asterisk replying to Session progress with an Ack then an Invite |
ASTERISK-19865: Forward a received 'answered elsewhere' |
ASTERISK-19866: Display pause reasont at cli queue show queuename |
ASTERISK-19867: asterisk fails to lower its priority when astcanary dies |
ASTERISK-19868: How to enable ExtensionState Check with SIP real time |
ASTERISK-19871: No translation path between various SILK sample rates |
ASTERISK-19874: Deadlock when voicemail ODBC database fails to respond. Asterisk does not respond to any sip requests. |
ASTERISK-19875: Behavior change in BLINDTRANSFER variable such that it is not available at the h extension |
ASTERISK-19876: app_voicemail: make_email_file() sends emails with localized Date header |
ASTERISK-19877: Queued Calls Remain In Queue When Phone In Queue Reboots |
ASTERISK-19880: Can't transcode between SILK codecs |
ASTERISK-19881: My Asterisk crashes multiple times daily |
ASTERISK-19882: Asterisk fails to unsubscribe from PubSub nodes when using ejabberd |
ASTERISK-19883: [patch] - RTP packet with Timestamp=0 on Multicast paging |
ASTERISK-19886: MOH class set by mohinterpret does not show up in NoOp(${CHANNEL(musicclass)}) |
ASTERISK-19887: Pattern matching broken on Local channels (ast 1.8) |
ASTERISK-19888: Choppy Audio from one client, great audio from two others, difference in RTP type number, why? |
ASTERISK-19889: Asterisk crashes due to memory corruption |
ASTERISK-19890: Crash occurs when using SIP realtime with SQL Server database, if a SIP client is un-registered |
ASTERISK-19891: Realtime queue problem with joinempty option |
ASTERISK-19892: If Asterisk sends a 481 to an initial INVITE that contained a to-tag, then Asterisk will not recognize the ensuing ACK |
ASTERISK-19894: Asterisk ooh323 channl has voice lost during established calls - log notes: No Open LogicalChannels |
ASTERISK-19896: safe_asterisk |
ASTERISK-19898: OriginateResponse event is received with Reason 3, 'remote end is ringing', while remote end destination is unreachable |
ASTERISK-19899: Confbridge user number announcement segfaults for number > 2 |
ASTERISK-19901: Asterisk 1.8.13.0 Blockers |
ASTERISK-19902: Asterisk 10.5.0 Blockers |
ASTERISK-19903: Remove AST_FLAG_ANSWERED_ELSEWHERE, duplicating the functionality of AST_CAUSE_ANSWERED_ELSEWHERE |
ASTERISK-19905: Security Vulnerability: remotely exploitable crash in chan_skinny if client is disconnected when client is not in on-hook state |
ASTERISK-19908: Add an ami function to refresh a voicemail box |
ASTERISK-19911: echocan_mode not documented |
ASTERISK-19912: [patch] Add ANI-2 / OLI reporting to chan_sip |
ASTERISK-19914: Incorrect SIP cause to Asterisk cause mapping in chan_sip |
ASTERISK-19915: CLIP - India |
ASTERISK-19916: Asterisk gets exhausted of all the file resources |
ASTERISK-19917: When WaitExten is called immediately on a new SIP call - DTMF fails to be detected with the RFC2833 and In-band methods, but succeeds with SIP INFO |
ASTERISK-19918: MoH (Music on Hold) is stopped after call in a queue is terminated |
ASTERISK-19919: Incorrect a=inactive when call changes from SIP_PAGE2_CALL_ONHOLD_INACTIVE to SIP_PAGE2_CALL_ONHOLD_ONEDIR |
ASTERISK-19920: res_adsi module is loaded (or Asterisk thinks it is) despite no modules.conf, noload or autoload=no instructions |
ASTERISK-19921: codec_dahdi: Wrong number of encoder/decoder channels. |
ASTERISK-19922: AudioHook mutex errors when Local channel is optimized away |
ASTERISK-19923: Asterisk crashing due to memory corruptions in chan_sip/voicemail |
ASTERISK-19924: Setting the variable CHANNEL(tonezone) when using AMI command Originate seems to have no effect |
ASTERISK-19925: Accountcode field in realtime does not work! |
ASTERISK-19934: [patch] dialplan reload context |
ASTERISK-19935: Patch to make app_system check if a command failed to execute due to permission denied. |
ASTERISK-19936: Segmentation fault with extensions.conf not present, empty or everything commented out |
ASTERISK-19937: Still not working - Failed to CANCEL a call in ringing state (SIP) in 1.8.9.2, 1.8.11, 1.8.12.2 |
ASTERISK-19939: Write test for the Asterisk Test Suite to cover subscribing for MWI in chan_sip |
ASTERISK-19941: Crash in res_config_ldap when used with realtime extensions |
ASTERISK-19942: SIP Session Timers do not honour refresher |
ASTERISK-19943: Ref leak in app_mixmonitor, manager_mixmonitor |
ASTERISK-19948: Asterisk 1.8 manager redirect command fails when redirecting multiple channels currently bridged together via dial command. |
ASTERISK-19949: app_meetme unable to write frame to Local/XXX channel (stuck channel) |
ASTERISK-19957: cdr_adaptative_odbc missing records |
ASTERISK-19960: Incorrect data in queue_log, event TRANSFER, field data1 |
ASTERISK-19961: Completely silenced tone zone |
ASTERISK-19962: Asterisk 1.8.12.0 can not play files in exten h |
ASTERISK-19963: Add AccountCode field to the manager hangup event |
ASTERISK-19965: Add IPv6 Support To Manager |
ASTERISK-19966: Masquerade super test fails when timing source is timer_fd |
ASTERISK-19968: TCP Session-Timers not dropping call |
ASTERISK-19969: Enhance astobj2 to support other types of containers. |
ASTERISK-19970: Add red-black tree container to astobj2. |
ASTERISK-19971: Segfault in realtime_multi_ldap (possible invalid arguments) |
ASTERISK-19974: AMI ORIGINATE - can't set CALLERID(num-pres) on outgoing call |
ASTERISK-19977: CLONE - Asterisk res ldap |
ASTERISK-19978: Gtalk Channel can't hangup on Android ICS 4.0.4 |
ASTERISK-19979: Request URI port inclusion inconsistency during outbound registration |
ASTERISK-19981: Dial plan variables are limited to 4K (4096 bytes) |
ASTERISK-19983: ConfBridge does not expose a mechanism to change the language on the Bridging channel, defaulting to 'en' |
ASTERISK-19984: sip configuration with insecure |
ASTERISK-19985: Bridge - not always returning to the right context,priority,extension as it should. |
ASTERISK-19986: meetme - Error writing file for recording |
ASTERISK-19987: Create chan_jingle2/res_xmpp Test Plan |
ASTERISK-19989: Add XMPP support to Asterisk Test Suite |
ASTERISK-19991: Memory leak in cel_pgsql |
ASTERISK-19992: SIP re-INVITEs have no transaction timeout |
ASTERISK-19994: app_voicemail should be able to decide on storage engines at runtime |
ASTERISK-19995: Recording calls is strongly degraded when using RTP packetization of 60 ms (g729: 60) |
ASTERISK-19996: Asterisk logs two CDR entries for a Local call. |
ASTERISK-19997: Faulty asterisk sip registrations - cause incoming network buffer to rise |